diff --git a/src/audio/module_adapter/module/waves.c b/src/audio/module_adapter/module/waves.c
index ed8a3f9c219b..89e1e6b02551 100644
--- a/src/audio/module_adapter/module/waves.c
+++ b/src/audio/module_adapter/module/waves.c
@@ -622,7 +622,7 @@ static int waves_codec_init(struct processing_module *mod)
 		struct module_config *setup_cfg = &waves_codec->setup_cfg;
 
 		/* allocate memory for set up config */
-		setup_cfg->data = rballoc(0, SOF_MEM_CAPS_RAM, codec->cfg.size);
+		setup_cfg->data = module_allocate_memory(mod, codec->cfg.size, 16);
 		if (!setup_cfg->data) {
 			comp_err(dev, "waves_codec_init(): failed to alloc setup config");
 			module_free_memory(mod, waves_codec);
@@ -634,6 +634,7 @@ static int waves_codec_init(struct processing_module *mod)
 		ret = memcpy_s(setup_cfg->data, setup_cfg->size, codec->cfg.data, setup_cfg->size);
 		if (ret) {
 			comp_err(dev, "waves_codec_init(): failed to copy setup config %d", ret);
+			module_free_memory(mod, setup_cfg->data);
 			module_free_memory(mod, waves_codec);
 			return ret;
 		}
diff --git a/src/audio/smart_amp/smart_amp.c b/src/audio/smart_amp/smart_amp.c
index e5f617d6f56a..eeea155a7f3d 100644
--- a/src/audio/smart_amp/smart_amp.c
+++ b/src/audio/smart_amp/smart_amp.c
@@ -25,8 +25,8 @@ DECLARE_TR_CTX(maxim_dsm_comp_tr, SOF_UUID(maxim_dsm_comp_uuid),
 #define SOF_SMART_AMP_MODEL 1
 
 typedef int(*smart_amp_proc)(struct comp_dev *dev,
-			     const struct audio_stream *source,
-			     const struct audio_stream *sink, uint32_t frames,
+			     const struct audio_stream __sparse_cache *source,
+			     const struct audio_stream __sparse_cache *sink, uint32_t frames,
 			     int8_t *chan_map, bool is_feedback);
 
 struct smart_amp_data {
@@ -516,8 +516,11 @@ static int smart_amp_trigger(struct comp_dev *dev, int cmd)
 	switch (cmd) {
 	case COMP_TRIGGER_START:
 	case COMP_TRIGGER_RELEASE:
-		if (sad->feedback_buf)
-			buffer_zero(sad->feedback_buf);
+		if (sad->feedback_buf) {
+			struct comp_buffer __sparse_cache *buf = buffer_acquire(sad->feedback_buf);
+			buffer_zero(buf);
+			buffer_release(buf);
+		}
 		break;
 	case COMP_TRIGGER_PAUSE:
 	case COMP_TRIGGER_STOP:
@@ -530,8 +533,8 @@ static int smart_amp_trigger(struct comp_dev *dev, int cmd)
 }
 
 static int smart_amp_process(struct comp_dev *dev,
-			     const struct audio_stream *source,
-			     const struct audio_stream *sink,
+			     const struct audio_stream __sparse_cache *source,
+			     const struct audio_stream __sparse_cache *sink,
 			     uint32_t frames, int8_t *chan_map,
 			     bool is_feedback)
 {
@@ -540,14 +543,14 @@ static int smart_amp_process(struct comp_dev *dev,
 
 	if (!is_feedback)
 		ret = smart_amp_ff_copy(dev, frames,
-					sad->source_buf, sad->sink_buf,
+					source, sink,
 					chan_map, sad->mod_handle,
 					sad->in_channels, sad->out_channels);
 	else
 		ret = smart_amp_fb_copy(dev, frames,
-					sad->feedback_buf, sad->sink_buf,
+					source, sink,
 					chan_map, sad->mod_handle,
-					sad->feedback_buf->stream.channels);
+					source->channels);
 	return ret;
 }
 
@@ -569,83 +572,70 @@ static smart_amp_proc get_smart_amp_process(struct comp_dev *dev)
 static int smart_amp_copy(struct comp_dev *dev)
 {
 	struct smart_amp_data *sad = comp_get_drvdata(dev);
-	struct comp_buffer *source_buf = sad->source_buf;
-	struct comp_buffer *sink_buf = sad->sink_buf;
+	struct comp_buffer __sparse_cache *source_buf = buffer_acquire(sad->source_buf);
+	struct comp_buffer __sparse_cache *sink_buf = buffer_acquire(sad->sink_buf);
 	uint32_t avail_passthrough_frames;
 	uint32_t avail_feedback_frames;
 	uint32_t avail_frames;
 	uint32_t source_bytes;
 	uint32_t sink_bytes;
 	uint32_t feedback_bytes;
-	int ret = 0;
 
 	comp_dbg(dev, "smart_amp_copy()");
 
-	source_buf = buffer_acquire(source_buf);
-	sink_buf = buffer_acquire(sink_buf);
-
 	/* available bytes and samples calculation */
-	avail_passthrough_frames =
-		audio_stream_avail_frames(&sad->source_buf->stream,
-					  &sad->sink_buf->stream);
-
-	source_buf = buffer_release(source_buf);
-	sink_buf = buffer_release(sink_buf);
+	avail_passthrough_frames = audio_stream_avail_frames(&source_buf->stream,
+							     &sink_buf->stream);
 
 	avail_frames = avail_passthrough_frames;
 
 	if (sad->feedback_buf) {
-		struct comp_buffer *buf = sad->feedback_buf;
+		struct comp_buffer __sparse_cache *feedback_buf = buffer_acquire(sad->feedback_buf);
 
-		buf = buffer_acquire(buf);
-		if (comp_get_state(dev, buf->source) == dev->state) {
+		if (comp_get_state(dev, feedback_buf->source) == dev->state) {
 			/* feedback */
 			avail_feedback_frames =
-				audio_stream_get_avail_frames(&buf->stream);
+				audio_stream_get_avail_frames(&feedback_buf->stream);
 
-			avail_frames = MIN(avail_passthrough_frames,
-					   avail_feedback_frames);
+			avail_feedback_frames = MIN(avail_passthrough_frames,
+						    avail_feedback_frames);
 
-			feedback_bytes = avail_frames *
-				audio_stream_frame_bytes(&buf->stream);
-
-			buffer_release(buf);
+			feedback_bytes = avail_feedback_frames *
+				audio_stream_frame_bytes(&feedback_buf->stream);
 
 			comp_dbg(dev, "smart_amp_copy(): processing %d feedback frames (avail_passthrough_frames: %d)",
-				 avail_frames, avail_passthrough_frames);
+				 avail_feedback_frames, avail_passthrough_frames);
 
 			/* perform buffer writeback after source_buf process */
-			buffer_stream_invalidate(sad->feedback_buf, feedback_bytes);
-			sad->process(dev, &sad->feedback_buf->stream,
-				     &sad->sink_buf->stream, avail_frames,
+			buffer_stream_invalidate(feedback_buf, feedback_bytes);
+			sad->process(dev, &feedback_buf->stream,
+				     &sink_buf->stream, avail_feedback_frames,
 				     sad->config.feedback_ch_map, true);
 
-			comp_update_buffer_consume(sad->feedback_buf, feedback_bytes);
-		} else {
-			buffer_release(buf);
+			comp_update_buffer_cached_consume(feedback_buf, feedback_bytes);
 		}
+
+		buffer_release(feedback_buf);
 	}
 
 	/* bytes calculation */
-	source_buf = buffer_acquire(source_buf);
 	source_bytes = avail_frames * audio_stream_frame_bytes(&source_buf->stream);
-	source_buf = buffer_release(source_buf);
-
-	sink_buf = buffer_acquire(sink_buf);
 	sink_bytes = avail_frames * audio_stream_frame_bytes(&sink_buf->stream);
-	sink_buf = buffer_release(sink_buf);
 
 	/* process data */
-	buffer_stream_invalidate(sad->source_buf, source_bytes);
-	sad->process(dev, &sad->source_buf->stream, &sad->sink_buf->stream,
+	buffer_stream_invalidate(source_buf, source_bytes);
+	sad->process(dev, &source_buf->stream, &sink_buf->stream,
 		     avail_frames, sad->config.source_ch_map, false);
-	buffer_stream_writeback(sad->sink_buf, sink_bytes);
+	buffer_stream_writeback(sink_buf, sink_bytes);
 
 	/* source/sink buffer pointers update */
-	comp_update_buffer_consume(sad->source_buf, source_bytes);
-	comp_update_buffer_produce(sad->sink_buf, sink_bytes);
+	comp_update_buffer_cached_consume(source_buf, source_bytes);
+	comp_update_buffer_cached_produce(sink_buf, sink_bytes);
 
-	return ret;
+	buffer_release(sink_buf);
+	buffer_release(source_buf);
+
+	return 0;
 }
 
 static int smart_amp_reset(struct comp_dev *dev)
@@ -666,7 +656,7 @@ static int smart_amp_reset(struct comp_dev *dev)
 static int smart_amp_prepare(struct comp_dev *dev)
 {
 	struct smart_amp_data *sad = comp_get_drvdata(dev);
-	struct comp_buffer *source_buffer;
+	struct comp_buffer __sparse_cache *source_c, *buf_c;
 	struct list_item *blist;
 	int ret;
 	int bitwidth;
@@ -679,40 +669,47 @@ static int smart_amp_prepare(struct comp_dev *dev)
 
 	/* searching for stream and feedback source buffers */
 	list_for_item(blist, &dev->bsource_list) {
-		source_buffer = container_of(blist, struct comp_buffer,
-					     sink_list);
-		source_buffer = buffer_acquire(source_buffer);
-		if (source_buffer->source->ipc_config.type == SOF_COMP_DEMUX)
+		struct comp_buffer *source_buffer = container_of(blist, struct comp_buffer,
+								 sink_list);
+		source_c = buffer_acquire(source_buffer);
+
+		if (source_c->source->ipc_config.type == SOF_COMP_DEMUX) {
 			sad->feedback_buf = source_buffer;
-		else
+		} else {
 			sad->source_buf = source_buffer;
-		source_buffer = buffer_release(source_buffer);
+			sad->in_channels = source_c->stream.channels;
+		}
+
+		buffer_release(source_c);
 	}
 
 	sad->sink_buf = list_first_item(&dev->bsink_list, struct comp_buffer,
 					source_list);
 
-	sad->in_channels = sad->source_buf->stream.channels;
-	sad->out_channels = sad->sink_buf->stream.channels;
+	buf_c = buffer_acquire(sad->sink_buf);
+	sad->out_channels = buf_c->stream.channels;
+	buffer_release(buf_c);
+
+	source_c = buffer_acquire(sad->source_buf);
 
 	if (sad->feedback_buf) {
-		struct comp_buffer *buf = sad->feedback_buf;
-
-		buf = buffer_acquire(buf);
-		buf->stream.channels = sad->config.feedback_channels;
-		buf->stream.rate = sad->source_buf->stream.rate;
-		buffer_release(buf);
-		ret = smart_amp_check_audio_fmt(sad->source_buf->stream.rate,
-						sad->source_buf->stream.channels);
+		buf_c = buffer_acquire(sad->feedback_buf);
+
+		buf_c->stream.channels = sad->config.feedback_channels;
+		buf_c->stream.rate = source_c->stream.rate;
+		buffer_release(buf_c);
+
+		ret = smart_amp_check_audio_fmt(source_c->stream.rate,
+						source_c->stream.channels);
 		if (ret) {
 			comp_err(dev, "[DSM] Format not supported, sample rate: %d, ch: %d",
-				 sad->source_buf->stream.rate,
-				 sad->source_buf->stream.channels);
+				 source_c->stream.rate,
+				 source_c->stream.channels);
 			goto error;
 		}
 	}
 
-	switch (sad->source_buf->stream.frame_fmt) {
+	switch (source_c->stream.frame_fmt) {
 	case SOF_IPC_FRAME_S16_LE:
 		bitwidth = 16;
 		break;
@@ -724,33 +721,33 @@ static int smart_amp_prepare(struct comp_dev *dev)
 		break;
 	default:
 		comp_err(dev, "[DSM] smart_amp_process() error: not supported frame format %d",
-			 sad->source_buf->stream.frame_fmt);
+			 source_c->stream.frame_fmt);
 		goto error;
 	}
-	if (sad->mod_handle->bitwidth != bitwidth)	{
-		sad->mod_handle->bitwidth = bitwidth;
-		comp_info(dev, "[DSM] Re-initialized for %d bit processing", bitwidth);
 
-		ret = smart_amp_init(sad->mod_handle, dev);
-		if (ret) {
-			comp_err(dev, "[DSM] Re-initialization error.");
-			goto error;
-		}
-		ret = maxim_dsm_restore_param(sad->mod_handle, dev);
-		if (ret) {
-			comp_err(dev, "[DSM] Restoration error.");
-			goto error;
-		}
+	sad->mod_handle->bitwidth = bitwidth;
+	comp_info(dev, "[DSM] Re-initialized for %d bit processing", bitwidth);
+
+	ret = smart_amp_init(sad->mod_handle, dev);
+	if (ret) {
+		comp_err(dev, "[DSM] Re-initialization error.");
+		goto error;
+	}
+	ret = maxim_dsm_restore_param(sad->mod_handle, dev);
+	if (ret) {
+		comp_err(dev, "[DSM] Restoration error.");
+		goto error;
 	}
 
 	sad->process = get_smart_amp_process(dev);
 	if (!sad->process) {
 		comp_err(dev, "smart_amp_prepare(): get_smart_amp_process failed");
 		ret = -EINVAL;
-		goto error;
 	}
 
 error:
+	buffer_release(source_c);
+
 	smart_amp_flush(sad->mod_handle, dev);
 	return ret;
 }
diff --git a/src/audio/smart_amp/smart_amp_generic.c b/src/audio/smart_amp/smart_amp_generic.c
index c9b79ff284d0..c969f66eadec 100644
--- a/src/audio/smart_amp/smart_amp_generic.c
+++ b/src/audio/smart_amp/smart_amp_generic.c
@@ -24,9 +24,9 @@ static void smart_amp_fb_generic(int32_t x)
 
 #if CONFIG_FORMAT_S16LE
 static void smart_amp_s16_ff_default(const struct comp_dev *dev,
-				     const struct audio_stream *source,
-				     const struct audio_stream *sink,
-				     const struct audio_stream *feedback,
+				     const struct audio_stream __sparse_cache *source,
+				     const struct audio_stream __sparse_cache *sink,
+				     const struct audio_stream __sparse_cache *feedback,
 				     uint32_t frames)
 {
 	int16_t *x;
@@ -52,9 +52,9 @@ static void smart_amp_s16_ff_default(const struct comp_dev *dev,
 
 #if CONFIG_FORMAT_S24LE
 static void smart_amp_s24_ff_default(const struct comp_dev *dev,
-				     const struct audio_stream *source,
-				     const struct audio_stream *sink,
-				     const struct audio_stream *feedback,
+				     const struct audio_stream __sparse_cache *source,
+				     const struct audio_stream __sparse_cache *sink,
+				     const struct audio_stream __sparse_cache *feedback,
 				     uint32_t frames)
 {
 	int32_t *x;
@@ -80,9 +80,9 @@ static void smart_amp_s24_ff_default(const struct comp_dev *dev,
 
 #if CONFIG_FORMAT_S32LE
 static void smart_amp_s32_ff_default(const struct comp_dev *dev,
-				     const struct audio_stream *source,
-				     const struct audio_stream *sink,
-				     const struct audio_stream *feedback,
+				     const struct audio_stream __sparse_cache *source,
+				     const struct audio_stream __sparse_cache *sink,
+				     const struct audio_stream __sparse_cache *feedback,
 				     uint32_t frames)
 {
 	int32_t *x;
@@ -106,9 +106,9 @@ static void smart_amp_s32_ff_default(const struct comp_dev *dev,
 
 #if CONFIG_FORMAT_S16LE
 static void smart_amp_s16_fb_default(const struct comp_dev *dev,
-				     const struct audio_stream *source,
-				     const struct audio_stream *sink,
-				     const struct audio_stream *feedback,
+				     const struct audio_stream __sparse_cache *source,
+				     const struct audio_stream __sparse_cache *sink,
+				     const struct audio_stream __sparse_cache *feedback,
 				     uint32_t frames)
 {
 	int16_t *x;
@@ -130,9 +130,9 @@ static void smart_amp_s16_fb_default(const struct comp_dev *dev,
 
 #if CONFIG_FORMAT_S24LE
 static void smart_amp_s24_fb_default(const struct comp_dev *dev,
-				     const struct audio_stream *source,
-				     const struct audio_stream *sink,
-				     const struct audio_stream *feedback,
+				     const struct audio_stream __sparse_cache *source,
+				     const struct audio_stream __sparse_cache *sink,
+				     const struct audio_stream __sparse_cache *feedback,
 				     uint32_t frames)
 {
 	int32_t *x;
@@ -154,9 +154,9 @@ static void smart_amp_s24_fb_default(const struct comp_dev *dev,
 
 #if CONFIG_FORMAT_S32LE
 static void smart_amp_s32_fb_default(const struct comp_dev *dev,
-				     const struct audio_stream *source,
-				     const struct audio_stream *sink,
-				     const struct audio_stream *feedback,
+				     const struct audio_stream __sparse_cache *source,
+				     const struct audio_stream __sparse_cache *sink,
+				     const struct audio_stream __sparse_cache *feedback,
 				     uint32_t frames)
 {
 	int32_t *x;
diff --git a/src/audio/smart_amp/smart_amp_maxim_dsm.c b/src/audio/smart_amp/smart_amp_maxim_dsm.c
index 73ff38c38b89..14cf31119f65 100644
--- a/src/audio/smart_amp/smart_amp_maxim_dsm.c
+++ b/src/audio/smart_amp/smart_amp_maxim_dsm.c
@@ -579,7 +579,7 @@ int smart_amp_check_audio_fmt(int sample_rate, int ch_num)
 }
 
 static int smart_amp_get_buffer(int32_t *buf, uint32_t frames,
-				struct audio_stream *stream,
+				const struct audio_stream __sparse_cache *stream,
 				int8_t *chan_map, uint32_t num_ch)
 {
 	int idx, ch;
@@ -629,7 +629,7 @@ static int smart_amp_get_buffer(int32_t *buf, uint32_t frames,
 }
 
 static int smart_amp_put_buffer(int32_t *buf, uint32_t frames,
-				struct audio_stream *stream,
+				const struct audio_stream __sparse_cache *stream,
 				int8_t *chan_map, uint32_t num_ch_in,
 				uint32_t num_ch_out)
 {
@@ -681,8 +681,8 @@ static int smart_amp_put_buffer(int32_t *buf, uint32_t frames,
 }
 
 int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames,
-		      struct comp_buffer *source,
-		      struct comp_buffer *sink, int8_t *chan_map,
+		      const struct audio_stream __sparse_cache *source,
+		      const struct audio_stream __sparse_cache *sink, int8_t *chan_map,
 		      struct smart_amp_mod_struct_t *hspk,
 		      uint32_t num_ch_in, uint32_t num_ch_out)
 {
@@ -703,12 +703,12 @@ int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames,
 	num_ch_out = MIN(num_ch_out, SMART_AMP_FF_OUT_MAX_CH_NUM);
 
 	ret = smart_amp_get_buffer(hspk->buf.frame_in,
-				   frames, &source->stream, chan_map,
+				   frames, source, chan_map,
 				   num_ch_in);
 	if (ret)
 		goto err;
 
-	switch (source->stream.frame_fmt) {
+	switch (source->frame_fmt) {
 	case SOF_IPC_FRAME_S16_LE:
 		maxim_dsm_ff_proc(hspk, dev,
 				  hspk->buf.frame_in,
@@ -728,7 +728,7 @@ int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames,
 	}
 
 	ret = smart_amp_put_buffer(hspk->buf.frame_out,
-				   frames, &sink->stream, chan_map,
+				   frames, sink, chan_map,
 				   MIN(num_ch_in, SMART_AMP_FF_MAX_CH_NUM),
 				   MIN(num_ch_out, SMART_AMP_FF_OUT_MAX_CH_NUM));
 	if (ret)
@@ -741,8 +741,8 @@ int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames,
 }
 
 int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames,
-		      struct comp_buffer *source,
-		      struct comp_buffer *sink, int8_t *chan_map,
+		      const struct audio_stream __sparse_cache *source,
+		      const struct audio_stream __sparse_cache *sink, int8_t *chan_map,
 		      struct smart_amp_mod_struct_t *hspk,
 		      uint32_t num_ch)
 {
@@ -762,12 +762,12 @@ int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames,
 	num_ch = MIN(num_ch, SMART_AMP_FB_MAX_CH_NUM);
 
 	ret = smart_amp_get_buffer(hspk->buf.frame_iv,
-				   frames, &source->stream,
+				   frames, source,
 				   chan_map, num_ch);
 	if (ret)
 		goto err;
 
-	switch (source->stream.frame_fmt) {
+	switch (source->frame_fmt) {
 	case SOF_IPC_FRAME_S16_LE:
 		maxim_dsm_fb_proc(hspk, dev, hspk->buf.frame_iv,
 				  frames * num_ch, sizeof(int16_t));
@@ -784,6 +784,6 @@ int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames,
 	return 0;
 err:
 	comp_err(dev, "[DSM] Not supported frame format : %d",
-		 source->stream.frame_fmt);
+		 source->frame_fmt);
 	return ret;
 }
diff --git a/src/include/sof/audio/smart_amp/smart_amp.h b/src/include/sof/audio/smart_amp/smart_amp.h
index 14e372f1122f..6438d7f082bc 100644
--- a/src/include/sof/audio/smart_amp/smart_amp.h
+++ b/src/include/sof/audio/smart_amp/smart_amp.h
@@ -136,9 +136,9 @@ struct smart_amp_mod_struct_t {
 };
 
 typedef void (*smart_amp_func)(const struct comp_dev *dev,
-			       const struct audio_stream *source,
-			       const struct audio_stream *sink,
-			       const struct audio_stream *feedback,
+			       const struct audio_stream __sparse_cache *source,
+			       const struct audio_stream __sparse_cache *sink,
+			       const struct audio_stream __sparse_cache *feedback,
 			       uint32_t frames);
 struct smart_amp_func_map {
 	uint16_t frame_fmt;
@@ -151,14 +151,14 @@ int smart_amp_init(struct smart_amp_mod_struct_t *hspk, struct comp_dev *dev);
 int smart_amp_flush(struct smart_amp_mod_struct_t *hspk, struct comp_dev *dev);
 /* Feed forward processing function */
 int smart_amp_ff_copy(struct comp_dev *dev, uint32_t frames,
-		      struct comp_buffer *source,
-		      struct comp_buffer *sink, int8_t *chan_map,
+		      const struct audio_stream __sparse_cache *source,
+		      const struct audio_stream __sparse_cache *sink, int8_t *chan_map,
 		      struct smart_amp_mod_struct_t *hspk,
 		      uint32_t num_ch_in, uint32_t num_ch_out);
 /* Feedback processing function */
 int smart_amp_fb_copy(struct comp_dev *dev, uint32_t frames,
-		      struct comp_buffer *source,
-		      struct comp_buffer *sink, int8_t *chan_map,
+		      const struct audio_stream __sparse_cache *source,
+		      const struct audio_stream __sparse_cache *sink, int8_t *chan_map,
 		      struct smart_amp_mod_struct_t *hspk,
 		      uint32_t num_ch);
 /* memory usage calculation for the component */
diff --git a/tools/topology/topology1/CMakeLists.txt b/tools/topology/topology1/CMakeLists.txt
index d4c652b4df14..15bfa738f759 100644
--- a/tools/topology/topology1/CMakeLists.txt
+++ b/tools/topology/topology1/CMakeLists.txt
@@ -149,6 +149,7 @@ set(TPLGS
 	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682\;-DCODEC=MAX98360A\;-DFMT=s32le\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-DBT_OFFLOAD\;-DSPK_MIC_PERIOD_US=10000"
 	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682-waves\;-DCODEC=MAX98360A\;-DFMT=s32le\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-DWAVES"
 	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682-2way\;-DCODEC=MAX98360A_TDM\;-DFMT=s32le\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-D2CH_2WAY\;-DBT_OFFLOAD"
+	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682-2way-pdm1\;-DCODEC=MAX98360A_TDM\;-DFMT=s32le\;-DDMIC_DAI_LINK_16k_PDM=STEREO_PDM1\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-D2CH_2WAY\;-DBT_OFFLOAD"
 	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682-4ch\;-DCODEC=MAX98360A_TDM\;-DFMT=s32le\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-D4CH_PASSTHROUGH\;-DBT_OFFLOAD"
 	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682-rtnr\;-DCODEC=MAX98360A\;-DFMT=s32le\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-DCHANNELS=2\;-DRTNR"
 	"sof-tgl-max98357a-rt5682\;sof-adl-max98360a-rt5682-rtnr-google-aec\;-DCODEC=MAX98360A\;-DFMT=s32le\;-DPLATFORM=adl\;-DLINUX_MACHINE_DRIVER=sof_rt5682\;-DAMP_SSP=1\;-DCHANNELS=2\;-DGOOGLE_RTC_AUDIO_PROCESSING\;-DRTNR\;-DNOHOTWORD\;-DNO16KDMIC"